It is known that the transmission of an audio signal, for example, radio broadcast, cable transmission, satellite transmission or recorded signals can be accomplished by converting the analog audio signal into a digital audio signal having a particular resolution. The digital signal is transmitted and upon reception is reconverted to an analog signal. One advantage of this technique is an increase in the signal-to-noise ratio, particularly during playback.
The bandwidth needed for the transmission of a digital signal is essentially determined by the number of scanning values per time unit which have to be transmitted, as well as by the resolution desired. Typically the transmission band width is kept as narrow as possible to enable the use of a narrow band channel, or to enable the transmission of a number of audio signals over an existing channel. The bandwidth needed can be minimized by reducing the scanning values or by reducing the number of bits per scanning value.
Typically, either of these reductions results in a reduction in the quality of the reproduction. In a known method of improving the playback quality, (described in German Patent DE OS 35 06 912.0) the digital audio signal is segmented into successive temporal segments and transformed into a short time spectrum which represents, for the respective time segments (for example, 20 ms) the spectral components of the signal. Because of psychoacoustic laws signal components which are not perceived by the listener, and which, therefore, are not needed to convey information, can usually be more easily found in the short time spectrum than in the time range. Such unneeded signal components are either less heavily weighted or completely left out of the transmission. The use of these measures permits a considerable portion of the unneeded data to be omitted from transmission and the average bit rate can be significantly reduced.
The method described by J. P. Princen and A. B. Bradley in "Analysis/Synthesis Filter-bank Design Based on Time Domain Aliasing Cancellation", IEEE Transactions Acoustics, Speech, Signal Processing, volume ASSP-34, pages 1153 through 1161, October 1986, is suitable for the partitioning of the signal into segments. This article describes a conversion technique in which overlapping blocks with rounded-off window functions are generated in the windows without additional coefficients in the frequency range. In this method N values are sampled from the input signal by means of a window function f(n) of length N, and subsequently converted into N/2 significant coefficients in the frequency range. The reconversion calculates N scanning values, which are again weighted using the window function f(n), from the N/2 coefficients.
However, the output signal of the reconversion differs from the input signal originally converted. The precise reconstruction of the input signal is only made possible when the output values of successive reconversions are added in the overlap area of N/2 scanning values. In order that the input signal can be recovered by means of this so-called "overlap-add" technique, the window function f(n) must comply with the following conditions: EQU f(N-1-n)=f(n)0&lt;=n&lt;=N-1 (1) EQU f.sup.2 (N/2-1-n)+f.sup.2 (n)=2 0&lt;=n&lt;=N/2-1 (2).
The first condition (1) corresponds to a symmetry of f(n). The second condition (2) corresponds to the point symmetry of the square of f(n) in one half of a window. Taking these conditions into consideration, the effective window length of the conversion can be varied between N/2 and N scanning values.
The choice of window length when using the partitioning method of conversion coding is an important consideration. A long window length with a shape which is as rounded-off as possible results in good frequency selectivity. However, the error will extend over the entire effective window length due to quantization of the coefficients after the reconversion. This can have a negative effect on the subjective quality of the coded signal, especially with large changes in the amplitude of the signal which is to be coded.
The choice of a shorter window causes a deterioration in the frequency selectivity, this has a negative effect on the conversion gain, particularly with strongly correlated input signals. In comparison, errors can be limited to the window concerned by quantizing the coefficients in case of large signal changes so that their effects on neighboring windows are avoided.